Merge tag 'sound-6.5-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Slightly bigger than I wished, but here we go, a collection of fixes
  for 6.5.

  The only change in the core side is the ease for repeated ASoC error
  messages, and the rest are all pretty device-specific small fixes
  (including regression fixes) for ASoC Intel and HD-audio / USB-audio
  quirks"

* tag 'sound-6.5-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/realtek - Remodified 3k pull low procedure
  ASoC: rt1308-sdw: fix random louder sound
  ALSA: hda/cs8409: Support new Dell Dolphin Variants
  ALSA: hda/realtek: Switch Dell Oasis models to use SPI
  ALSA: hda/realtek: Add quirks for HP G11 Laptops
  ASoC: meson: axg-tdm-formatter: fix channel slot allocation
  ASoC: SOF: ipc4-topology: Update the basecfg for copier earlier
  ASoC: SOF: intel: hda: Clean up link DMA for IPC3 during stop
  ASoC: Intel: sof-sdw-cs42142: fix for codec button mapping
  ASoC: Intel: sof-sdw: update jack detection quirk for LunarLake RVP
  ASoC: SOF: Fix incorrect use of sizeof in sof_ipc3_do_rx_work()
  ASoC: lower "no backend DAIs enabled for ... Port" log severity
  ASoC: rt5665: add missed regulator_bulk_disable
  ASoC: max98363: don't return on success reading revision ID
  ALSA: usb-audio: Add support for Mythware XA001AU capture and playback interfaces.
  ASoC: fsl: micfil: Use dual license micfil code

17 files changed, 125 insertions, 46 deletions

diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c
index b288874e401e..36b411d1a960 100644
--- a/sound/pci/hda/patch_cs8409-tables.c
+++ b/sound/pci/hda/patch_cs8409-tables.c
@@ -550,6 +550,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN),
 	SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN),
 	SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN),
+	SND_PCI_QUIRK(0x1028, 0x0C73, "Dolphin", CS8409_DOLPHIN),
+	SND_PCI_QUIRK(0x1028, 0x0C75, "Dolphin", CS8409_DOLPHIN),
+	SND_PCI_QUIRK(0x1028, 0x0C7D, "Dolphin", CS8409_DOLPHIN),
+	SND_PCI_QUIRK(0x1028, 0x0C7F, "Dolphin", CS8409_DOLPHIN),
 	{} /* terminator */
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 44fccfb93cff..dc7b7a407638 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9422,11 +9422,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0cbd, "Dell Oasis 13 CS MTL-U", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1028, 0x0cbe, "Dell Oasis 13 2-IN-1 MTL-U", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1028, 0x0cbf, "Dell Oasis 13 Low Weight MTU-L", ALC245_FIXUP_CS35L41_SPI_2),
-	SND_PCI_QUIRK(0x1028, 0x0cc1, "Dell Oasis 14 MTL-H/U", ALC287_FIXUP_CS35L41_I2C_2),
-	SND_PCI_QUIRK(0x1028, 0x0cc2, "Dell Oasis 14 2-in-1 MTL-H/U", ALC287_FIXUP_CS35L41_I2C_2),
-	SND_PCI_QUIRK(0x1028, 0x0cc3, "Dell Oasis 14 Low Weight MTL-U", ALC287_FIXUP_CS35L41_I2C_2),
-	SND_PCI_QUIRK(0x1028, 0x0cc4, "Dell Oasis 16 MTL-H/U", ALC287_FIXUP_CS35L41_I2C_2),
-	SND_PCI_QUIRK(0x1028, 0x0cc5, "Dell Oasis MLK 14 RPL-P", ALC287_FIXUP_CS35L41_I2C_2),
+	SND_PCI_QUIRK(0x1028, 0x0cc1, "Dell Oasis 14 MTL-H/U", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1028, 0x0cc2, "Dell Oasis 14 2-in-1 MTL-H/U", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1028, 0x0cc3, "Dell Oasis 14 Low Weight MTL-U", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1028, 0x0cc4, "Dell Oasis 16 MTL-H/U", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -9617,7 +9616,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED),
-	SND_PCI_QUIRK(0x103c, 0x8c26, "HP HP EliteBook 800G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c48, "HP EliteBook 860 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c49, "HP Elite x360 830 2-in-1 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -10638,6 +10643,7 @@ static int patch_alc269(struct hda_codec *codec)
 	spec = codec->spec;
 	spec->gen.shared_mic_vref_pin = 0x18;
 	codec->power_save_node = 0;
+	spec->en_3kpull_low = true;
 
 #ifdef CONFIG_PM
 	codec->patch_ops.suspend = alc269_suspend;
@@ -10720,14 +10726,16 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->shutup = alc256_shutup;
 		spec->init_hook = alc256_init;
 		spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */
-		if (codec->bus->pci->vendor == PCI_VENDOR_ID_AMD)
-			spec->en_3kpull_low = true;
+		if (codec->core.vendor_id == 0x10ec0236 &&
+		    codec->bus->pci->vendor != PCI_VENDOR_ID_AMD)
+			spec->en_3kpull_low = false;
 		break;
 	case 0x10ec0257:
 		spec->codec_variant = ALC269_TYPE_ALC257;
 		spec->shutup = alc256_shutup;
 		spec->init_hook = alc256_init;
 		spec->gen.mixer_nid = 0;
+		spec->en_3kpull_low = false;
 		break;
 	case 0x10ec0215:
 	case 0x10ec0245:
diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c
index b5c69bba0e48..2dfaf4fcfbd3 100644
--- a/sound/soc/codecs/max98363.c
+++ b/sound/soc/codecs/max98363.c
@@ -185,10 +185,10 @@ static int max98363_io_init(struct sdw_slave *slave)
 	pm_runtime_get_noresume(dev);
 
 	ret = regmap_read(max98363->regmap, MAX98363_R21FF_REV_ID, &reg);
-	if (!ret) {
+	if (!ret)
 		dev_info(dev, "Revision ID: %X\n", reg);
-		return ret;
-	}
+	else
+		goto out;
 
 	if (max98363->first_hw_init) {
 		regcache_cache_bypass(max98363->regmap, false);
@@ -198,10 +198,11 @@ static int max98363_io_init(struct sdw_slave *slave)
 	max98363->first_hw_init = true;
 	max98363->hw_init = true;
 
+out:
 	pm_runtime_mark_last_busy(dev);
 	pm_runtime_put_autosuspend(dev);
 
-	return 0;
+	return ret;
 }
 
 #define MAX98363_RATES SNDRV_PCM_RATE_8000_192000
diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c
index f43520ca3187..e566c8ddd3e9 100644
--- a/sound/soc/codecs/rt1308-sdw.c
+++ b/sound/soc/codecs/rt1308-sdw.c
@@ -52,6 +52,7 @@ static bool rt1308_volatile_register(struct device *dev, unsigned int reg)
 	case 0x300a:
 	case 0xc000:
 	case 0xc710:
+	case 0xcf01:
 	case 0xc860 ... 0xc863:
 	case 0xc870 ... 0xc873:
 		return true;
@@ -213,7 +214,7 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave)
 {
 	struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev);
 	int ret = 0;
-	unsigned int tmp;
+	unsigned int tmp, hibernation_flag;
 
 	if (rt1308->hw_init)
 		return 0;
@@ -242,6 +243,10 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave)
 
 	pm_runtime_get_noresume(&slave->dev);
 
+	regmap_read(rt1308->regmap, 0xcf01, &hibernation_flag);
+	if ((hibernation_flag != 0x00) && rt1308->first_hw_init)
+		goto _preset_ready_;
+
 	/* sw reset */
 	regmap_write(rt1308->regmap, RT1308_SDW_RESET, 0);
 
@@ -282,6 +287,12 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave)
 	regmap_write(rt1308->regmap, 0xc100, 0xd7);
 	regmap_write(rt1308->regmap, 0xc101, 0xd7);
 
+	/* apply BQ params */
+	rt1308_apply_bq_params(rt1308);
+
+	regmap_write(rt1308->regmap, 0xcf01, 0x01);
+
+_preset_ready_:
 	if (rt1308->first_hw_init) {
 		regcache_cache_bypass(rt1308->regmap, false);
 		regcache_mark_dirty(rt1308->regmap);
diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index 83c367af91da..525713c33d71 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -4472,6 +4472,8 @@ static void rt5665_remove(struct snd_soc_component *component)
 	struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component);
 
 	regmap_write(rt5665->regmap, RT5665_RESET, 0);
+
+	regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies);
 }
 
 #ifdef CONFIG_PM
diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index 3f08082a55be..9d01225dedd9 100644
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -1,4 +1,4 @@
-// SPDX-License-Identifier: GPL-2.0
+// SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause
 // Copyright 2018 NXP
 
 #include <linux/bitfield.h>
@@ -1254,4 +1254,4 @@ module_platform_driver(fsl_micfil_driver);
 
 MODULE_AUTHOR("Cosmin-Gabriel Samoila <cosmin.samoila@nxp.com>");
 MODULE_DESCRIPTION("NXP PDM Microphone Interface (MICFIL) driver");
-MODULE_LICENSE("GPL v2");
+MODULE_LICENSE("Dual BSD/GPL");
diff --git a/sound/soc/fsl/fsl_micfil.h b/sound/soc/fsl/fsl_micfil.h
index 9237a1c4cb8f..fee9fe3d9119 100644
--- a/sound/soc/fsl/fsl_micfil.h
+++ b/sound/soc/fsl/fsl_micfil.h
@@ -1,4 +1,4 @@
-/* SPDX-License-Identifier: GPL-2.0 */
+/* SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause */
 /*
  * PDM Microphone Interface for the NXP i.MX SoC
  * Copyright 2018 NXP
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index dbee8c98ff01..0201029899ca 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -476,7 +476,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "Lunar Lake Client Platform"),
 		},
-		.driver_data = (void *)(RT711_JD2_100K),
+		.driver_data = (void *)(RT711_JD2),
 	},
 	{}
 };
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c
index c4a16e4c9f69..ad130d913415 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l42.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c
@@ -99,9 +99,9 @@ static int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd)
 	jack = &ctx->sdw_headset;
 
 	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
-	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
-	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
-	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
 
 	ret = snd_soc_component_set_jack(component, jack, NULL);
 
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 9883dc777f63..63333a2b0a9c 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -30,27 +30,32 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map,
 					struct axg_tdm_stream *ts,
 					unsigned int offset)
 {
-	unsigned int val, ch = ts->channels;
-	unsigned long mask;
-	int i, j;
+	unsigned int ch = ts->channels;
+	u32 val[AXG_TDM_NUM_LANES];
+	int i, j, k;
+
+	/*
+	 * We need to mimick the slot distribution used by the HW to keep the
+	 * channel placement consistent regardless of the number of channel
+	 * in the stream. This is why the odd algorithm below is used.
+	 */
+	memset(val, 0, sizeof(*val) * AXG_TDM_NUM_LANES);
 
 	/*
 	 * Distribute the channels of the stream over the available slots
-	 * of each TDM lane
+	 * of each TDM lane. We need to go over the 32 slots ...
 	 */
-	for (i = 0; i < AXG_TDM_NUM_LANES; i++) {
-		val = 0;
-		mask = ts->mask[i];
-
-		for (j = find_first_bit(&mask, 32);
-		     (j < 32) && ch;
-		     j = find_next_bit(&mask, 32, j + 1)) {
-			val |= 1 << j;
-			ch -= 1;
+	for (i = 0; (i < 32) && ch; i += 2) {
+		/* ... of all the lanes ... */
+		for (j = 0; j < AXG_TDM_NUM_LANES; j++) {
+			/* ... then distribute the channels in pairs */
+			for (k = 0; k < 2; k++) {
+				if ((BIT(i + k) & ts->mask[j]) && ch) {
+					val[j] |= BIT(i + k);
+					ch -= 1;
+				}
+			}
 		}
-
-		regmap_write(map, offset, val);
-		offset += regmap_get_reg_stride(map);
 	}
 
 	/*
@@ -63,6 +68,11 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map,
 		return -EINVAL;
 	}
 
+	for (i = 0; i < AXG_TDM_NUM_LANES; i++) {
+		regmap_write(map, offset, val[i]);
+		offset += regmap_get_reg_stride(map);
+	}
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 8896227e4fb7..3aa6b988cb4b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -38,6 +38,7 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd,
 	switch (ret) {
 	case -EPROBE_DEFER:
 	case -ENOTSUPP:
+	case -EINVAL:
 		break;
 	default:
 		dev_err(rtd->dev,
@@ -2466,8 +2467,11 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream)
 
 	/* there is no point preparing this FE if there are no BEs */
 	if (list_empty(&fe->dpcm[stream].be_clients)) {
-		dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n",
-				fe->dai_link->name);
+		/* dev_err_once() for visibility, dev_dbg() for debugging UCM profiles */
+		dev_err_once(fe->dev, "ASoC: no backend DAIs enabled for %s, possibly missing ALSA mixer-based routing or UCM profile\n",
+			     fe->dai_link->name);
+		dev_dbg(fe->dev, "ASoC: no backend DAIs enabled for %s\n",
+			fe->dai_link->name);
 		ret = -EINVAL;
 		goto out;
 	}
diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c
index f3513796c189..f33051eac1c0 100644
--- a/sound/soc/sof/intel/hda-dai-ops.c
+++ b/sound/soc/sof/intel/hda-dai-ops.c
@@ -372,6 +372,7 @@ static const struct hda_dai_widget_dma_ops hda_ipc4_chain_dma_ops = {
 static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai,
 				 struct snd_pcm_substream *substream, int cmd)
 {
+	struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream);
 	struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
 
 	switch (cmd) {
@@ -379,9 +380,17 @@ static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *c
 	case SNDRV_PCM_TRIGGER_STOP:
 	{
 		struct snd_sof_dai_config_data data = { 0 };
+		int ret;
 
 		data.dai_data = DMA_CHAN_INVALID;
-		return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data);
+		ret = hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data);
+		if (ret < 0)
+			return ret;
+
+		if (cmd == SNDRV_PCM_TRIGGER_STOP)
+			return hda_link_dma_cleanup(substream, hext_stream, cpu_dai);
+
+		break;
 	}
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_PAUSE, NULL);
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 3297dea493aa..863865f3d77e 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -107,9 +107,8 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai
 	return sdai->platform_private;
 }
 
-static int hda_link_dma_cleanup(struct snd_pcm_substream *substream,
-				struct hdac_ext_stream *hext_stream,
-				struct snd_soc_dai *cpu_dai)
+int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream,
+			 struct snd_soc_dai *cpu_dai)
 {
 	const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, cpu_dai);
 	struct sof_intel_hda_stream *hda_stream;
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 3f7c6fb05e5d..5b9e4ebcc18b 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -963,5 +963,7 @@ const struct hda_dai_widget_dma_ops *
 hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget);
 int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags,
 		   struct snd_sof_dai_config_data *data);
+int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream,
+			 struct snd_soc_dai *cpu_dai);
 
 #endif
diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c
index 2c5aac31e8b0..580960ff273d 100644
--- a/sound/soc/sof/ipc3.c
+++ b/sound/soc/sof/ipc3.c
@@ -1001,7 +1001,7 @@ void sof_ipc3_do_rx_work(struct snd_sof_dev *sdev, struct sof_ipc_cmd_hdr *hdr,
 
 	ipc3_log_header(sdev->dev, "ipc rx", hdr->cmd);
 
-	if (hdr->size < sizeof(hdr) || hdr->size > SOF_IPC_MSG_MAX_SIZE) {
+	if (hdr->size < sizeof(*hdr) || hdr->size > SOF_IPC_MSG_MAX_SIZE) {
 		dev_err(sdev->dev, "The received message size is invalid: %u\n",
 			hdr->size);
 		return;
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index a4e1a70b607d..11361e1cd688 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1731,6 +1731,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
 
 	*ipc_config_size = ipc_size;
 
+	/* update pipeline memory usage */
+	sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config);
+
 	/* copy IPC data */
 	memcpy(*ipc_config_data, (void *)copier_data, sizeof(*copier_data));
 	if (gtw_cfg_config_length)
@@ -1743,9 +1746,6 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
 		       gtw_cfg_config_length,
 		       &ipc4_copier->dma_config_tlv, dma_config_tlv_size);
 
-	/* update pipeline memory usage */
-	sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config);
-
 	return 0;
 }
 
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index efb4a3311cc5..5d72dc8441cb 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -4507,6 +4507,35 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+{
+	/* Advanced modes of the Mythware XA001AU.
+	 * For the standard mode, Mythware XA001AU has ID ffad:a001
+	 */
+	USB_DEVICE_VENDOR_SPEC(0xffad, 0xa001),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "Mythware",
+		.product_name = "XA001AU",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_IGNORE_INTERFACE,
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE,
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE,
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 #undef USB_DEVICE_VENDOR_SPEC
 #undef USB_AUDIO_DEVICE